Designing & Deploying UC networks with Cisco Session Management Edition

Designing & Deploying UC networks with Cisco Session Management Edition BRKUCC-2931 Follow us on Twitter for real time updates of the event: @ciscoliveeurope, #CLEUR

Housekeeping We value your feedback- don't forget to complete your online session evaluations after each session & the Overall Conference Evaluation which will be available online from Thursday Visit the World of Solutions and Meet the Engineer Visit the Cisco Store to purchase your recommended readings Please switch off your mobile phones After the event don t forget to visit Cisco Live Virtual: www.ciscolivevirtual.com Follow us on Twitter for real time updates of the event: @ciscoliveeurope, #CLEUR BRKUCC-2931 2012 Cisco and/or its affiliates. All rights reserved. Cisco Public 2

Session Objectives At the end of this session you should : Have a good understanding of how to design SME deployments Understand the benefits of SME designs Understand how to size an SME cluster SME DEPLOYMENT GUIDE The content of this presentation is based on the SME Deployment Guide which can be found here : http://www.cisco.com/en/us/products/ps10661/products_implementati on_design_guides_list.html 3

Agenda What is a Session Management Edition (SME) cluster? Trunk Aggregation and Trunk Design recommendations Call Re-routing in SME deployments SME Deployments Secure Trunks Encrypted Signaling and Media Dial Plan Aggregation and Number Transformation Call Admission Control within SME based UC networks Signaling Delay considerations - Regional SME clusters QSIG in SME Deployments Mobility and SME UC Applications and SME SME Deployments with Unified Contact Centre Enterprise Sizing the SME cluster Summary 4

What is a Session Management Edition cluster? H323 Trunk MGCP Trunk SIP Trunk A CUCM cluster and SME cluster use exactly the same software A CUCM cluster is typically used to register 10,000s of Phones An SME cluster is typically used as a platform for Trunk and Dial Plan aggregation Both CUCM and SME support Voice, Video and Encrypted calls Support for SME deployments was introduced with UC version 7.1(2) UC version 8.5 introduces a number of features that enhance SME functionality : Improved SIP Trunk and H323 Inter Cluster Trunk functionality Improved through-cluster routing SIP Normalization and Transparency scripts 5

Why deploy Session Management Edition? An SME cluster at the core of your network allows you to flexibly manage your UC system as it grows and changes. PSTN Voicemail System Conferencing System Unified CM Session Management Edition Cluster H323 MGCP SIP CUCM Clusters with H323 QSIG Trunks to SME PBXs/ CUCM Clusters with SIP Trunks to SME CUCM Clusters with H323 Trunks to SME PBXs with MGCP QSIG Trunks to SME PBXs with MGCP Q931 Gateway Trunks CUCM/PBX with SIP QSIG Trunks to SME Leaf Unified CM Clusters/ Leaf UC Systems An SME cluster can interconnect 1000 s of UC systems using SIP, H323, or MGCP Trunks SME allows you to reduce UC system complexity by centralizing your dial plan and call routing rules in the SME cluster which in turn allows you to simplify the dial plan and management of the connected UC systems 6

Reasons for deploying an SME cluster PSTN Voicemail System Conferencing System Unified CM Session Management Edition Cluster H323 MGCP SIP CUCM Clusters with H323 QSIG Trunks to SME PBXs/ CUCM Clusters with SIP Trunks to SME CUCM Clusters with H323 Annex M1 Trunks to SME PBXs with MGCPTrunks to SME PBXs with MGCP Q931 Gateway Trunks CUCM/PBXs with SIP QSIG Trunks to SME A centralized Dial Plan in SME allows you to : Globalize and Normalize the called and calling numbers used by all Leaf systems Manage overlapping number ranges in Leaf systems Configure find me call routing using Route List and Route Groups Re-route calls via the PSTN when the device cannot be reached via and IP path SME Mobility features allow you to offer Single Number Reach functionality to devices on 3 rd Party UC systems SME Normalization scripts allow you to modify any inbound or outbound SIP message and SDP body content simplifying interoperability with 3 rd party UC systems 7

SME cluster - PBX aggregation SME migration to a CUCM cluster PSTN PSTN PSTN PSTN PSTN PSTN PSTN Phase 1 PBX Aggregation PSTN Phase 2 Centralized IP PSTN PSTN Phase 3 PBX decommissioning Phase 4 CUCM cluster 8

Trunk Aggregation and Trunk Design recommendations for SME deployments

Trunk Aggregation and Trunk Design recommendations Summary deployment recommendations for SME Trunks Inter Cluster Trunk feature comparison New Trunk features that simplify SME designs Load Balancing, availability and redundancy SME Trunks to IOS gateways and 3 rd Party UC systems CUCM Leaf cluster Trunks UC version design considerations SME Clustering Over the WAN design considerations 10

Summary design and deployment recommendations for SME Trunks UC version 8.5+ is recommended for SME clusters as it introduces a range new Trunking features that simplify SME deployments UC version 8.5+ is preferred for CUCM Leaf clusters as this allows them to use the new Trunking features in this release, but earlier versions are also supported With Leaf CUCM cluster UC version 8.5+ SIP ICTs and Trunks are preferred With Leaf CUCM cluster UC versions prior to 8.5 H323 ICTs are preferred Although MGCP Trunks to gateways offer benefits in terms of ease of dial plan configuration they do not (cannot) benefit from the SIP and H323 Trunk features introduced with UC version 8.5 11

Unified CM 8.5 - Inter Cluster Trunks SIP Trunks vs H.323 Trunks Feature Comparison Support for + character Signaling Authentication and Encryption Media Encryption Run On All Nodes feature Up to 16 destination addresses feature Calling Line ID / Name Presentation / Restriction Connected Line ID / Name Presentation / Restriction OPTIONS Ping ilbc, AAC, ISAC and G.Clear Support G.711, G.722, G.723, G.729 Support SIP Subscribe / Notify, Publish Presence QSIG Path Replacement QSIG Call Completion No Reply / Busy Subscriber Topology Aware - RSVP Based Call Admission Control Message Waiting Indicator (On /Off) Video / T.38 Fax support Legend: Yes Limited support H.323 (Q.SIG) SIP (QSIG) 12 No TLS

New Trunk features that simplify SME designs UC version 8.5 introduces the following Trunk features : H323 Inter Cluster Trunks o Run On All Unified CM Nodes o Up to 16 Destination IP addresses SIP Trunks and SIP Inter Cluster Trunks o Run On All Unified CM Nodes o Up to 16 Destination IP addresses o SIP Options Ping o SIP Normalization Scripts Route Lists o Run On All Unified CM Nodes In conjunction with the Route Local feature these new features greatly simplify Leaf CUCM cluster and SME cluster Trunk design and deployment 13

New Trunk features that simplify SME designs Run On All Unified CM Nodes Up to 16 Destination IP addresses These features : o Reduce the number of configured Trunks o Simplify call distribution o Simplify call routing through Leaf and SME clusters by taking advantage of the Route Local rule.. SIP/H323 ICT Trunk SIP Trunk A CUB E SIP/H323 ICT Trunk SIP Trunk A CUB E SIP Trunk B CUB E CUB E CUCM SME CUCM SME 14

New Trunk features that simplify SME designs Outbound SIP Trunks, H323 Inter Cluster Trunks and Route Lists can take advantage of the Route Local Rule by using the Run On All Unified CM Nodes feature The Route Local Rule If the CUCM node that the inbound call arrives on also has an instance of the selected outbound trunk for that call then use this node to onward route the call The Route Local rule reduces (and can eliminate) call set up traffic between CUCM nodes with a cluster SIP/H323 ICT Trunk SIP Trunk CUB E SIP/H323 ICT Trunk SIP Trunk A CUB E SIP Trunk B CUB E CUB E Route List 15

UC 8.5 SIP Trunk Feature - OPTIONS Ping SIP ICT Trunk SIP Trunk OPTIONS Ping is activated on a per SIP Trunk basis Each node running the SIP Trunk daemon in the originating cluster uses OPTIONS Ping to determine the availability of each defined destination IP address - CUCM will not attempt to establish a call to an unavailable remote peer - SIP Trunk - In Service whilst one remote peer is reachable - SIP Trunk - Out Of Service state when all remote peers are unreachable - CUCM 8.5 SIP Trunks Dynamic reachability detection - Pre CUCM 8.5 Trunks - Per call time out 16

Unified CM Trunks Load Balancing, Availability & Redundancy Summary UC 8.5 Features HA Features affecting Calls originating from a CUCM cluster HA Features for Trunk destinations Calls over single Trunks Calls over multiple Trunks Route Lists and Route Groups Up to 16 Destination IP Addresses H323 ICTs and SIP Trunks Call Manager Groups Call Manager Groups H323 Gatekeeper DNS SRV SIP Trunks Run on All Nodes H323 ICTs and SIP Trunks Route Local Run on All Nodes H323 ICTs, SIP Trunks and Route Groups Route Local SIP OPTIONS Ping SIP Trunks only 17

UC 8.5 SIP Trunks Normalization Scripts SIP/SCCP SIP Trunk Script Normalization allows incoming and outgoing SIP messages to be modified on their way through a CUCM SIP Trunk. SIP Trunk H323 Trunk SIP Trunk Script SIP Trunk IP PSTN The Normalization feature is designed to improve interoperability between CUCM SIP Trunks and SIP based 3rd Party SIP PBXs, Applications & IP PSTN services. MGCP Trunk Script SIP Trunk IP PSTN Normalization is independent of what the SIP Trunk connects to on the other side of CUCM. e.g. Normalization Script Remove display name from SIP Header function fixinboundpai() local pai = Sip.getHeader("P-Asserted-Identity") local displayname = getdisplayname(pai) local uri = geturi(pai) local number = getuserpart(uri) if displayname == number then Sip.modifyHeader("P-Asserted-Identity", uri) SIP Skinny H.323 MGCP SIP Trunk calls SIP Trunk calls SIP Trunk calls SIP Trunk calls Normalization uses a scripting environment to allow customers to modify SIP messages and SDP content on a per trunk basis. Scripting Guide at : http://www.cisco.com/en/us/docs/voice_ip_comm/cucm/sip_tn/8_5_1/sip_t_n.html 18

UC 8.5 SIP Trunks Transparency Scripts SIP Trunk SIP Trunk Script Transparency allows CUCM to pass headers, parameters, and SDP content from one SIP call leg to the other. SIP Trunk SIP Trunk SIP Trunk Script SIP Trunk Script Transparency Script Sip.allowHeader( A-Callid ) Sip.allowHeader( A-ConversationId ) function A.inbound_INVITE Sip.passThroughHeader( A-Callid ) Sip.passThroughHeader( A-ConversationId ) Sip.passThroughHeaderValue("Supported", "xnortel-sipvc") Sip.passThroughUriParameters("From", "uriparm1") Sip.passThroughHeaderParameters("From", "hparm1", "hparm2") end The Transparency feature is designed to improve the operation of and interoperability between 3rd Party SIP PBXs and Applications connected via CUCM/SME. Transparent pass through is only applicable when the call through CUCM is from SIP Trunk to SIP Trunk. SIP Trunk SIP Trunk calls Transparency uses the same scripting environment as Normalization to allow customers to pass SIP messages through CUCM. Transparency and Normalization features can be combined. Scripting Guide at : http://www.cisco.com/en/us/docs/voice_ip_comm/cucm/sip_tn/8_5_1/sip_t_n.html 19

SIP Trunks vs H323 and MGCP Trunks to gateways SIP Trunks support the Run On All Unified CM Nodes and Up to 16 destination IP addresses features H323 and MGCP Trunks to gateways and 3 rd party UC systems support standard Call Manager Groups and a single destination IP address Using standard Call Manager Groups (rather than Run on All Nodes) increases call set up traffic between nodes within a cluster Multiple Trunks may be required with the single destination IP address limitation Note MGCP Trunks are only active on one node in the Call Manager Group (as the signaling channel is back hauled to CUCM) H323 ICT Trunk H323 Trunk A SIP ICT Trunk MGCP Trunk A H323 Trunk B MGCP Trunk B Route List Route List Selected outbound Trunk Selected outbound Trunk 20

IOS Gateway Trunks SIP Trunks, H.323 Gateways, MGCP Gateways Feature Comparison Centralized Provisioning QSIG Tunneling Centralized CDR (DS0 Granularity in Unified CM CDR) MLPP (Preemption) H.323 SIP MGCP Hook-flash Transfer with Unified CM ISDN Overlap Sending NFAS SRTP (Unified CM to GW) CUCM 8.5 Run On All Nodes feature CUCM 8.5 Up to 16 destination addresses feature Mobility Manager VXML-Based Voice Profile Mgmt OPTIONS Ping TCL/VXML Apps (e.g. for CVP Integration) Voice & Data Integrated Access Fractional PRI TDM Variations: A-DID, E&M, PRI NFAS, CAMA, T1 FGD ISDN Video Switching on GW Set numbering Plan Type of Outgoing Calls G.Clear (Clear Channel Data) Support H.320 Video No GK 3 Active Nodes in a CMG 1 active Node in a GMG Workaround Legend: Yes Limited support No CMG Call Manager Group 21

SME Trunk Design Considerations Multi-Cluster Designs with CUCM 8.5 Leaf Clusters SIP Delayed Offer SIP IP PSTN SIP Early Offer SIP DO to EO CUCM 8.5 CUCM 8.5 CUCM 8.5 SME SIP ICT Trunks - Voice, Video and Encryption supported OPTIONS Ping, SIP Delayed Offer, Run on All Nodes, Multiple Destination Addresses, QSIG over SIP. (Call Back & Path Replacement) SIP IP PSTN Trunks Typically Voice only Early Offer usually required by SP OPTIONS Ping, SIP Delayed Offer, If EO required - provides SIP DO to EO, Run on All Nodes, Multiple Destination Addresses. (EO should be sent by the SP) SIP to 3 rd Party UC Systems Typically Voice Video & Encryption also supported OPTIONS Ping, SIP Delayed Offer, Run on All Nodes, If required - QSIG over SIP, If end device is capable it should send Early Offer. 22

SME Trunk Design Considerations Multi-Cluster Designs with Pre 8.5 CUCM Leaf Clusters SIP Delayed Offer SIP Early Offer H323 Slow Start SIP H323 IP PSTN SIP DO to EO CUCM 8.0 CUCM 7.X CUCM 8.5 SME Pre 8.5 Clusters - H323 ICT Trunks - Voice, Video and Encryption supported H323 Slow Start, Call Manager Groups, 3 Destination Addresses per Trunk, QSIG over H323. SIP IP PSTN Trunks Typically Voice only Early Offer usually required by SP OPTIONS Ping, SIP Delayed Offer, If EO required - provides SIP DO to EO, Run on All Nodes, Multiple Destination Addresses. (EO should be sent by the SP) SIP to 3rd Party UC Systems Typically Voice Video & Encryption also supported OPTIONS Ping, SIP Delayed Offer, Run on All Nodes, If required QSIG over SIP, If end device is capable it should send Early Offer. 23

SIP Trunk Design Considerations Multi-Cluster Designs Mixed Leaf Cluster CUCM Versions SIP Delayed Offer SIP Early Offer H323 Slow Start SIP H323 IP PSTN SIP DO to EO CUCM 8.0 CUCM 8.5 CUCM 8.5 SME Pre 8.5 Leaf Clusters H323 ICT Trunks Voice, Video and Encryption supported H323 Slow Start, Call Manager Groups, 3 Destination Addresses per Trunk, QSIG over H323. 8.5 Leaf Clusters SIP ICT Trunks Voice, Video and Encryption supported SIP Delayed Offer, OPTIONS Ping, Run on All Nodes, Multiple Destination Addresses, QSIG over SIP. SIP IP PSTN Trunks Typically Voice only Early Offer usually required by SP OPTIONS Ping, SIP Delayed Offer, If EO required provides SIP DO to EO, Run on All Nodes, Multiple Destination Addresses. (EO should be sent by the SP) 24

SIP Trunk Design Considerations Clustering Over the WAN Calls from Leaf Clusters to SME SIP Delayed Offer IP PSTN COW Bandwidth requirements as per the UC SRND - 80mS RTT San Jose West Coast DC New York Primary Secondary CUCM 8.5 SME 8.5 CUCM 8.5 San Francisco East Coast DC SIP CUCM 8.5 IP PSTN SIP DO to EO In each Leaf Cluster : Create and prioritize multiple SIP Trunks in Route Lists to distribute calls to each group of SME nodes in each Data Centre, Run Route Lists on all Nodes Enable Run on all Nodes on each Leaf Cluster SIP Trunk. Define destination IP addresses per Trunk for geographic call distribution Use SIP Delayed Offer (DO) on all Trunks (reduces MTP usage) For IP PSTNs - If Early Offer (EO) is required Use SIP DO to EO feature 25

SIP Trunk Design Considerations Clustering Over the WAN Calls from SME Cluster to Leafs SIP Delayed Offer IP PSTN COW Bandwidth requirements as per the UC SRND - 80mS RTT San Jose West Coast DC New York West West CUCM 8.5 SME 8.5 CUCM 8.5 San Francisco East Coast DC SIP CUCM 8.5 In the SME Cluster : Use standard Call Manager Groups for each SIP Trunk Define destination IP addresses for every call processing node in the Leaf cluster For routes to each Leaf cluster - Add SIP Trunks to a Route Group and Route List. Run Route Lists on all Nodes. Use Local Route Groups to route outbound calls over Trunks in the same data centre as the inbound Trunk call Use SIP Delayed Offer (DO) on all Trunks (reduces MTP usage) For IP PSTNs - If Early Offer (EO) is required Use SIP DO to EO feature 26 IP PSTN SIP DO to EO

Call Re-Routing in SME deployments

Call Re-routing in SME Deployments Single outbound SME Trunk No Circuit Available x SME IP PSTN IP PSTN SME x Call Admission Control Out Of Bandwidth Call Progress failure messages (e.g. No circuit, Number Unavailable, No bandwidth etc) are passed back to the originating Leaf cluster which can then re-route the call 28

Call Re-routing in SME Deployments Multiple outbound SME Trunks in a Route List IP PSTN No Circuit Available x SME IP PSTN x SME x Call Admission Control Out Of Bandwidth Call Progress failure messages (e.g. No circuit, Number Unavailable, No bandwidth etc ) are passed back to the SME cluster which can then re-route the call 29

Secure Trunks Encrypted Media and Signaling

SME Deployments Secure Trunks Encrypted Signaling and Media Secure Signaling and encrypted Media can be configured for SIP and H323 Trunks SIP Trunk security using TLS is preferred over H323 - H323 uses IPSEC and requires IPSEC Tunnels to be set up in the network rather than on CUCM/SME nodes Using central Certificate Authority with SIP Trunks reduces configuration overhead and simplifies certificate management Media Encryption (SRTP) can be set up without secure signaling but in this case security keys are sent in the clear 31

Secure SIP Trunks using TLS and an external CA Certificate Authority (CA) CA Certificate CA Signed CUCM Certificate Leaf Cluster A Encrypted Media SIP Inter Cluster Trunk using TLS for signaling encryption Session Management Edition Cluster Leaf Cluster B 32

Secure H323 Trunks using router based IPSec Tunnels Leaf Cluster A H323 Inter Cluster Trunk using router based IPSEC for signaling encryption Encrypted Media IPSEC Tunnel Session Management Edition Cluster Leaf Cluster B 33

Dial Plan Aggregation and Number Transformation

Dial Plan Aggregation and Number Transformation Basic Principles Number Transformation options Dial Plan and dialing habit variations Number Transformation capabilities by Trunk protocol type 35

SME Dial Plan Aggregation Basic Principles In the SME cluster a unique globalized dial plan should be implemented Inbound number globalization and outbound number localization can be implemented in the SME cluster Leaf Cluster SME Cluster Leaf UC system In Leaf clusters/ Leaf UC systems a localized dial plan may be implemented and users may have local dialing habits 36

SME - Number Normalization Tools Inbound number globalization and outbound number localization can be implemented in the SME cluster SME Cluster With Session Management Edition and CUCM, normalisation of calling and called party information can be achieved by using a variety of tools : Transformation calling search spaces on trunk or device pool level Transformations on route patterns Transformations on translation patterns Transformations on route lists This presentation focuses on number transformations at the Trunk level... But first lets look at some variations on dial plan and dialling habit implementations... 37

Internal Directory Numbers, +E.164 SME Dial Plan, Abbreviated On Net Dialing Calling Number +44-404-444-4000 Called Number +55-505-555-5000 Calling Number +44-404-444-4000 Called Number +55-505-555-5000 Cluster 1 DNs 444-4XXX SME Cluster + E.164 Dial Plan Cluster 2 DNs 555-5XXX Calling Number +44-404-444-4000 Called Number 8-555-5000 Calling Number +44-404-444-4000 Called Number 555-5000 Caller (444-4000) dials 8-555-5000 Leaf Cluster 1 Internal Directory Numbers (XXX-YYYY) Abbreviated Dialing (8-XXX-YYYY) Leaf Cluster 2 Internal Directory Numbers (XXX-YYYY) Abbreviated Dialing (8-XXX-YYYY) 38

Internal Directory Numbers, +E.164 SME Dial Plan, Abbreviated On Net Dialing All number transformations performed on SME cluster Calling Number +44-404-444-4000 Called Number +55-505-555-5000 Calling Number 8-444-4000 Called Number 555-5000 Cluster 1 DNs 444-4XXX SME Cluster + E.164 Dial Plan Cluster 2 DNs 555-5XXX Calling Number 444-4000 Called Number 8-555-5000 Calling Number 8-444-4000 Called Number 555-5000 Caller (444-4000) dials 8-555-5000 Leaf Cluster 1 Internal Directory Numbers (XXX-YYYY) Abbreviated Dialing (8-XXX-YYYY) Leaf Cluster 2 Internal Directory Numbers (XXX-YYYY) Abbreviated Dialing (8-XXX-YYYY) 39

+E.164 Directory Numbers, +E.164 SME Dial Plan, E.164 Dialing Calling Number +44-404-444-4000 Called Number +55-505-555-5000 Calling Number +44-404-444-4000 Called Number +55-505-555-5000 Cluster 1 DNs +44-404-444-4XXX SME Cluster + E.164 Dial Plan Cluster 2 DNs +55-505-555-5XXX Calling Number +44-404-444-4000 Called Number +55-505-555-5000 Calling Number +44-404-444-4000 Called Number +55-505-555-5000 Caller (+44-404-444-1000) dials 55-505-555-5000 Leaf Cluster 1 + E.164 Directory Numbers E.164 Dialing Leaf Cluster 2 + E.164 Directory Numbers E.164 Dialing 40

SIP Trunks Inbound and Outbound Digit manipulation capabilities Inbound SIP Trunk Outbound SIP Trunk SME /CUCM Cluster Inbound Calling Search Space AAR Calling Search Space Prefix DN Prefix Digits to Incoming Called Number Incoming Calling Number Prefix, Strip Transform Digits + char carried in SIP + char carried in called and calling number QSIG IEs Outbound Called Party Transformation CSS Outbound Calling Party Transformation CSS Caller ID DN Overwrite/Prefix Digits to Outbound Calling Number + char carried in SIP + char carried in called and calling number QSIG IEs 41

H323 Trunks and Gateways Inbound and Outbound Digit manipulation capabilities Inbound H323 Trunk Outbound H323 Trunk SME /CUCM Cluster Inbound Calling Search Space (CSS) AAR Calling Search Space Prefix DN Prefix Digits to Incoming Called Number Incoming Calling Number Prefix, Strip Transform Digits Incoming Called Number Prefix, Strip Transform Digits + char not carried in H323 + char not carried in QSIG Outbound Called Party Transformation CSS Outbound Calling Party Transformation CSS Caller ID DN Overwrite/Prefix Digits to Outbound Calling Number + char not carried in H323 + char not carried in QSIG 42

MGCP Trunks Inbound and Outbound Digit manipulation capabilities Inbound MGCP Trunk Outbound MGCP Trunk SME /CUCM Cluster Inbound Calling Search Space AAR Calling Search Space Prefix DN Prefix Digits to Incoming Called Number Incoming Calling Number Prefix, Strip Transform Digits + char not carried in Q.931 + char not carried in QSIG Outbound Called Party Transformation CSS Outbound Calling Party Transformation CSS Caller ID DN Overwrite/Prefix Digits to Outbound Calling Number + char not carried in Q.931 + char not carried in QSIG 43

Call Admission Control (CAC) in SME based deployments

Call Admission Control within SME based UC networks CAC Options : Locations based CAC RSVP based CAC Combinations of RSVP and Locations based CAC Local RSVP can be deployed in the SME cluster only - This has advantages in deployments where partially meshed international WAN circuits are used to interconnect SME and Leaf clusters, or where sites are dual homed to the WAN - RSVP can be used to accurately and efficiently use available WAN bandwidth 45

Call Admission Control within SME based UC networks Both Locations based Call Admission Control (CAC) and RVSP based CAC can be used in Leaf clusters and the SME cluster.. Locations based CAC for campus clusters provided by SME Locations CAC for Leaf clusters and SME using hub none locations Locations CAC for Leaf clusters and SME using phantom locations Local RSVP in the SME cluster Local RSVP in Leaf and SME clusters End to End RSVP in Leaf and SME clusters Mixed End to End RSVP in Leaf and SME clusters and Locations based CAC in other Leaf clusters 46

SME cluster provides Locations based CAC for campus clusters SME 8.5 Leaf 1 Leaf 3 SIP Trunk Leaf 2 LOCATION 1 Associated to Trunks for Leaf 1 and Leaf 2 (both Leaf Clusters have endpoints located in the same physical site) Registered Endpoint Location Association LOCATION 2 Associated to Leaf 3 Trunk (Leaf 3 is a single site cluster deployment) 47

Multi Branch Leaf clusters provide Locations CAC ICTs in Location Hub_None Hub_None ICT-CL1 ICT-CL2 Hub-None Hub_None ICT-SME SME Hub_None ICT-SME Central Site Cluster 1 Central Site Cluster 2 Cluster 1 Locations CAC 80k Deduction for Location B WAN Cluster 2 Locations CAC 80k Deduction for Location F Location B - 80 kbps Branch 1 Cluster 1 SIP Trunk Call Signaling Location Association Media Branch 2 Cluster 2 Location F - 80 kbps 48

Multi Branch Leaf clusters - Locations CAC Call Transfer Hub_None ICT-CL1 Hub_None ICT-CL2 Hub_None ICT-SME SME Hub_None ICT-SME Central Site Cluster 1 Central Site Cluster 2 Cluster 1 Locations CAC 160k Deduction for Location B (2 Calls) WAN Cluster 2 Call Transferred to Cluster 1 Location B Transfer Location B - 160 kbps Branch 1 Cluster 1 SIP Trunk Call Signaling Location Association Media Branch 2 Cluster 2 Location F No WAN calls 49

Multi Branch Leaf clusters provide Locations based CAC Phantom Locations allow the originating location of each call to be sent over ICTs Phantom ICT-SME Phantom ICT-CL1 SME Phantom ICT-CL2 1 2 Phantom ICT-SME ICTs in Location Phantom Central Site Cluster 1 Central Site Cluster 2 Cluster 1 Locations CAC 80k Deduction for Location B WAN Cluster 2 Locations CAC 80k Deduction for Location F Location B - 80 kbps Branch 1 Cluster 1 SIP Trunk Call Signaling Location Association Media Branch 2 Cluster 2 Location F - 80 kbps 50

Multi Branch Leaf clusters - Locations CAC Call Transfer Phantom Locations allow the originating location of each call to be sent over ICTs Hub_None ICT-CL1 Hub_None ICT-CL2 1 2 The originating location of the inbound transferred call is returned to cluster 1 Hub_None ICT-SME SME Hub_None ICT-SME Central Site Cluster 1 4 3 Central Site Cluster 2 Cluster 1 Locations CAC 0 kbps Deduction for Location B (0 WAN Calls) WAN Cluster 2 Call Transferred to Cluster 1 Location B Transfer Location B No WAN calls Branch 1 Cluster 1 SIP Trunk Call Signaling Location Association Media Branch 2 Cluster 2 Location F No WAN calls 51

SME cluster Deploying Local RSVP CAC for partially meshed WANs 10 WAN Link UC Bandwidth IP PSTN 100 10 10 1 1 10 10 For CUCM Leaf Clusters Using Locations based Call Admission Control What bandwidth value should be used for calls over SIP inter cluster Trunks into the WAN? For Voice Gateways Using Call Counting based Call Admission Control What value should be used for calls over SIP inter cluster Trunks into the WAN? Local RSVP in the SME cluster has advantages in deployments where partially meshed international WAN circuits are used, or where sites are dual homed to the WAN. RSVP does not need to be implemented in the leaf clusters 52

SME cluster Local RSVP CAC Locations Based CAC Local RSVP Local RSVP SME 8.5 SCCP H323 Inter cluster Trunk RSVP Agents registered to Leaf and SME clusters may be colocated on the same platform Leaf Cluster 1 7.X or below Location Leaf 1 Inter Location RSVP Policy = Mandatory Location Leaf 2 Leaf Cluster 2 7.X or below SME manages RSVP Agents between Clusters Local RSVP in the SME cluster has advantages in deployments where partially meshed international WAN circuits are used, or where sites are dual homed to the WAN. RSVP does not need to be implemented in the leaf clusters 53

SME cluster Local RSVP CAC Inter cluster Call Setup Locations Based CAC Local RSVP Local RSVP SME 8.5 RSVP Agents registered to Leaf and SME clusters may be colocated on the same platform Call Setup 1 3 Call Setup Leaf Cluster 1 7.X or below Location Leaf 1 2 Location Leaf 2 Leaf Cluster 2 7.X or below SME RSVP Reservation Call Signaling SCCP H323 Inter cluster Trunk RSVP Leaf RSVP Resv. With Local RSVP in the SME cluster Call set up proceeds on the outbound SME Trunk once the RSVP Reservation has succeeded between the two SME controlled RSVP agents 54

SME cluster End to End RSVP CAC with SIP Preconditions End-to-End RSVP with SIP Preconditions SME 8.5 SCCP SIP Inter cluster Trunk RSVP Leaf Cluster 1 8.0 or above Location Leaf 1 Location Leaf 2 Leaf Cluster 2 8.0 or above SIP Preconditions passed over SIP Inter cluster Trunks from Cluster 1 via SME to Cluster 2 SME RSVP Reservation 55

SME cluster End-to-End RSVP with SIP Preconditions extended into one Leaf cluster Locations Based CAC SCCP H323 Inter cluster Trunk SIP Inter cluster Trunk Local RSVP and End-to-End RSVP with SIP Preconditions SME 8.5 Leaf Cluster 1 7.X or below Location Leaf 1 Inter Location RSVP Policy = Mandatory Location Leaf 2 Leaf Cluster 2 8.0 or above SME RSVP Reservation 56

SME cluster Local RSVP CAC Co-Locating RSVP agents on the same router platform Local RSVP SME 8.5 Location Leaf 2 SME and Leaf cluster RSVP Agents co-located on the same router platform Local RSVP RSVP Agents registered to Leaf and SME clusters may be colocated on the same platform Leaf Cluster 2 sccp local GigabitEthernet0/0 sccp ccm 10.10.10.10 identifier 1 version 8.0 sccp ccm 10.10.20.20 identifier 2 version 7.0 sccp ip precedence 3 sccp sccp ccm group 10 associate ccm 1 priority 1 associate profile 10 register RSVP_Agent_SME sccp ccm group 20 associate ccm 2 priority 1 associate profile 20 register RSVP_Agent_Leaf2 dspfarm profile 10 mtp codec g729r8 codec pass-through rsvp maximum sessions software 300 associate application SCCP dspfarm profile 20 mtp codec g729r8 codec pass-through rsvp maximum sessions software 300 associate application SCCP 57

Signaling Delay Considerations Regional SME Clusters

Signaling Delay and SME based UC networks SME Recommendations for media delay are well defined (ITU Recommendation G.114. < 150mS = accptable, 150 400mS = acceptable with some impact on quality, > 400mS generally unacceptable) Recommendations for signaling delay are not well defined Primarily because the incurred delays are protocol dependent and the impact of long delay generally affects call set up rather than overall voice quality 59

Impact of Signaling Delay on Call Set Up in SME networks Messages exchanged before the caller hears ringback tone One way signaling Delay INVITE 100 Trying 180 Ringing 200 OK w/ SDP (Offer) ACK w/ SDP (Answer) SME INVITE 100 Trying 180 Ringing 200 OK w/ SDP (Offer) ACK w/ SDP (Answer) Delay before the caller hears the called user after ringback stops Two Way Media Messages exchanged before called user hears the caller after picking up their handset The diagram above shows an example of call set up delays and their impact on the users experience. (Note Phone to Call Agent signaling delay has been assumed to be minimal) Delays during call set up will vary based on the protocol(s) used, the trunk configuration and call agent operation making it difficult to calculate the time taken to establish each stage of the call set up. In most cases, signaling delays do not noticeably affect user experience. If signaling delays are a concern, consider deploying regional SME clusters. 60

Regional SME clusters IP PST N IP PST N SME America SME Europe SME Asia Pac IP PST N Provide a Regionalized SME cluster if there are multiple leaf UC systems in a geographic region with significant amounts of intra region traffic If regionalized SME clusters are not deployed, situate your SME cluster(s) closest to those leaf UC systems that generate the most inter cluster traffic 61

QSIG and SME deployments

QSIG and SME UC deployments With UC 8.5 - QSIG is supported over SIP, H323 and MGCP Trunks QSIG specific Features : o Call Back on Busy o Call Back on No Answer o Path Replacement o Calling Name o Connected Name QSIG features such as Call Back and Path Replacement can use calling and called numbers to determine whether the feature should be invoked. Numbers carried in QSIG APDUs are not modified by CUCM number transforms. In SME designs where number normalization is deployed Use the following CUCM/ SME service parameters : o Call Back : o Path Replacement : Connection Retention (default setting) PINX ID Use a globally unique number for the PINX ID of each cluster (The PINX ID is equivalent to any routable number in the UC system and uniquely identifies a cluster) 63

Benefits of QSIG Path Replacement in SME based UC deployments IP PSTN QSIG over SIP QSIG over H323 Media 1000 1001 9 10 4000 10 9 Phone 1000 calls Phone 4000 Signaling Call leg from 1XXX cluster via SME to 4XXX cluster Call Admission Control deducts the bandwidth for a call from 1XXX to 4XXX RTP media path direct between the two Phones over the WAN 64

Benefits of QSIG Path Replacement in SME based UC deployments IP PSTN QSIG over SIP QSIG over H323 Media 1000 1001 8 10 4000 10 8 Phone 4000 Transfers the call to Phone 1001 New Signaling Call leg from 4XXX cluster via SME to 1XXX cluster Call Admission Control deducts the bandwidth for a call from 4XXX to 1XXX Resulting Media Path direct between Phone 1000 and Phone 1001 65

Benefits of QSIG Path Replacement in SME based UC deployments IP PSTN SCCP/SIP Media 1000 1001 10 10 4000 10 10 QSIG Path Replacement for Tromboned calls Clears down redundant signaling Releases unused Call Admission Control bandwidth For more info on QSIG and SME deployments see The SME Deployment Guide : http://www.cisco.com/en/us/products/ps10661/products_implementation_design_guides_list.html and the CUCM Admin Guide : http://www.cisco.com/en/us/docs/voice_ip_comm/cucm/admin/8_6_1/ccmsys/a08procl.html#wp1139777 66

Cisco Mobility and SME deployments

Mobility Features in SME based UC networks SIP Delayed Offer SIP SIP Early Offer H323 H323 Slow Start IP PSTN TDM PSTN Unified Mobility User Unified Mobility User CUCM 8.0 CUCM 8.5 SME PBX with SNR capability For SME based UC designs - Unified Mobility can be deployed as follows : Standard Leaf CUCM cluster mobility deployment where all mobility features and PSTN access are provided on the Leaf cluster Leaf cluster mobility with SME based PSTN access 3 rd Party PBX Mobility/ Single Number Reach support with PSTN access via SME Mobility features provided by SME for a connected 3 rd party PBX PBX using SME based Mobility features 68

Leaf cluster mobility with SME based PSTN access IP PST N TDM PSTN Note SME PSTN access can be TDM or IP based Unified Mobility User Unified Mobility User CUCM 8.0 SME Supported Features Mobile Connect (aka SNR) Mobile Voice Access (MVA) Desk Phone pickup via hangup at mobile device and resume at desk phone. Remote Destination pickup using "Send Call to Mobile" Cisco Unified Mobile Communicator (CUMC)* Dual-mode Phones and Clients / Direct Connect (Cisco Jabber) Mobile Clients / BlackBerry MVS Unsupported Features - Features that use DTMF tones for feature invocation : Enterprise Feature Access (EFA) Desk Phone and Remote Destination pickup Mid-call Supplementary Services * EOL 16th Dec 2011 Migration Path to Cisco Jabber or MVS CUCM 8.5 69

Mobility features provided by SME for a 3 rd party PBX IP PST N TDM PSTN Note SME PSTN access can be TDM or IP based Unified Mobility User Unified Mobility User PBX with SNR capability SME PBX using SME based Mobility features Where a 3 rd Party PBX does not natively support any mobility features, the SME cluster can act as a mobility feature proxy by creating a Remote Destination Profile and two or more Remote Destination numbers/ IDs per user. One Remote Destination will be the directory number of the user s 3 rd Party PBX Phone and an additional Remote Destination number can be the user s mobile phone number. Supported mobility features : Mobile Connect (aka SNR) Mobile Voice Access (MVA) Unsupported Features - Features that use DTMF tones for feature invocation : Enterprise Feature Access (EFA) Desk Phone and Remote Destination pickup Mid-call Supplementary Services 70

UC Applications in SME deployments

UC Applications in SME deployments Voicemail System Conferencing System Cisco Unified Contact Center CER Cisco Emergency Responder Session Management Edition Cluster H323 MGCP SIP Cisco Unified Presence Server Operator Console SME supports the following centralized applications : Unity, Unity Connection Meeting Place, Meeting Place Express SIP and H323 based Video Conferencing systems 3 rd Party Voice Mail systems Fax servers Applications that track user or phone state must connect to the leaf cluster : Unified Contact Centre, Unified Contact Centre Express Cisco Unified Presence Server Attendant Console Cisco Emergency Responder 72

Centralized Unity VM in SME deployments Voicemail System Conferencing System Cisco Unified Contact Center CER Cisco Emergency Responder Session Management Edition Cluster H323 MGCP SIP Cisco Unified Presence Server Operator Console On Trunks between leaf clusters and SME and on Trunk connections to your Voicemail application ensure that the original called party/ redirecting number is sent with calls routed to voicemail. For Non QSIG Trunks Original Called Party /Redirecting number transport can be enabled by : Enabling inbound and outbound Redirecting Diversion Header Delivery on SIP Trunks Enabling inbound and outbound Redirecting Number IE Delivery on MGCP Gateways, H.323 Gateways and H.323 Trunks For QSIG enabled SIP, MGCP and H323 Trunks the Original Called Party number is sent in QSIG Diverting Leg Information APDUs. Note - The + character is not sent with the Diversion information sent in QSIG APDUs. For more information see the SME Deployment Guide : http://www.cisco.com/en/us/products/ps10661/products_implementation_design_guides_list.html 73

SME Deployments with Extension Mobility Cross Cluster IP PSTN Unified CM Session Management Edition Cluster EMCC Trunks Inter Cluster Trunks EMCC can be deployed as an overlay between leaf clusters in an SME based UC network. Leaf cluster EMCC Trunks do not point to the SME cluster. In each Leaf cluster, an EMCC Trunk(s) and its corresponding Remote Cluster destination information is defined such that each EMCC Trunk can reach all other EMCC enabled leaf clusters directly. 74

SME deployments with Cisco Unified Contact Centre

SME Deployments with UCCE Unified CM SME can be deployed in Unified Contact Center Enterprise (CCE) designs, where one or more leaf clusters are contact center-enabled. There are specific requirements for the architecture used in these scenarios : - Unified CCE is only certified using SIP as the protocol interconnecting the solution components. - The preferred DTMF transport in Unified CCE is RFC2833 ( inband DTMF ), since it is the only method supported for self-service applications on Unified CVP - Unified CCE and Unified CVP often use SIP headers to pass information between systems. By default, SME does not transparently relay some of these SIP headers and payloads from one trunk to another. However, if required, a transparency script can be applied to the SME trunk. - See the SME Deployment Guide for more details 76

Back Office CUCM s/pbx s SUPPORTED DESIGN Unified CM Session Management Edition Cluster Calls Transferred to Agents Contact Centre CUCM Cluster Back-office calls JTAPI IP PSTN Contact Centre calls CUSP Inbound calls to Contact Centre flow through CUSP and are routed to a Queue (e.g. On VXML gateway) via CUSP Queued Calls Calls transferred from the Queue to an Agent are routed through SME and optionally CUSP Calls to Back Office Users are routed via and SME No Calls are transferred from Back Office to Contact Centre VXML Gateway 77

Back Office CUCM s/pbx s NOT SUPPORTED Contact Centre CUCM Cluster Unified CM Session Management Edition Cluster JTAPI IP PSTN NOT SUPPORTED Queued Calls Inbound calls to Contact Centre flow through SME and are routed to a Queue (e.g. On VXML gateway) VXML Gateway 78

IPv6 based SME deployments

IPv6 based SME Deployments PSTN Voicemail System Conferencing System Unified CM Session Management Edition Cluster SIP Cisco UC products supporting IPv4 and IPv6 : - CUCM/SME SIP Trunks - Newer SCCP based Phones - SIP based IOS gateways - Cisco Unified Communications Manager Express and - SCCP based VG224 Analog gateways and Router FXS ports, - IOS MTPs - SIP Trunks to Unity Connection Dual Stack (IPv4 and IPv6) configuration is recommended Dual Stack ANAT is recommended for SIP Trunks http://www.cisco.com/en/us/docs/voice_ip_comm/cucm/srnd/ipv6/ipv6srnd.html 80

Centralized and Distributed PSTN

Centralized PSTN PSTN Voicemail System Conferencing System Unified CM Session Management Edition Cluster H323 MGCP SIP Centralized IP PSTN deployments are becoming increasingly popular as savings can be made through bulk call minute deals with Service Providers and by eliminating TDM circuits in branches Bear in mind that with no WAN connectivity no PSTN calls can be made or received A number of analog PSTN ports are often provided in the branch for back up and emergency outbound calls For emergency calls to the centralized PSTN, MLPP can be used to pre-empt existing WAN calls if insufficient CAC bandwidth is available 82

Distributed PSTN Voicemail System Conferencing System Unified CM Session Management Edition Cluster H323 MGCP SIP PSTN PSTN PSTN PSTN PSTN PSTN Typically TDM based branch PSTN circuits Distributed IP PSTN deployments growing Enables Tail End Hop Off Can be combined with centralized outbound PSTN on SME or centralized outbound PSTN on Leaf clusters which allows (for example, in country) TEHO to be deployed 83

SME Cluster Sizing

SME Cluster Sizing Summary Guidance User Traffic information required to size your SME cluster Average User Busy Hour Call Attempts (BHCA) Average Call Holding Time (CHT) % Off Net Traffic (Off Net to PSTN via SME) % On Net Traffic % Intra cluster Traffic Work with your Cisco account team or Cisco partner to size your SME cluster using the SME sizing tool http://tools.cisco.com/cucst/faces/login.jsp PSTN Voicemail System Conferencing System Unified CM Session Management Edition Cluster H323 MGCP SIP CUCM Clusters with H323 QSIG Trunks to SME PBXs/ CUCM Clusters with SIP Trunks to SME CUCM Clusters with H323 Annex M1 Trunks to SME PBXs with MGCPTrunks to SME PBXs with MGCP Q931 Gateway Trunks CUCM/PBXs with SIP QSIG Trunks to SME 85

Summary

Summary SME deployments allow you to simplify the UC network edge by aggregating dial plan, services, applications and PSTN access on an SME cluster SME based UC aggregation simplifies the management of your UC network as it grows and changes For design guidance see the SME Deployment Guide : http://www.cisco.com/en/us/products/ps10661/product s_implementation_design_guides_list.html 87

Additional Slides

SIP Trunk Signaling and Basic Operation SIP Messaging - Delayed and Early Offer SIP Early Offer Information about the calling device s media characteristics are sent with its initial SIP INVITE message The media characteristics are contained in the Session Description Protocol (SDP) body sent with the SIP INVITE The Offer in the SDP body will contain the IP Address, UDP Port number, list of codecs etc. supported by the calling device The called device selects which of the offered codecs it wishes to use for the call and returns its Answer in the SDP body of a SIP response The Answer also contains the IP address and UDP port number etc of the called device Once the Answer has been received and acknowledged two way media can be established Early Offer is widely used. INVITE w/ SDP (Offer) 100 Trying 180 Ringing 200 OK w/ SDP (Answer) ACK Two Way Media INVITE w/ SDP (Offer) 100 Trying 180 Ringing 200 OK w/ SDP (Answer) ACK 89

SIP Trunk Signaling and Basic Operation SIP Messaging - Delayed and Early Offer SIP Delayed Offer No information about the calling device s media characteristics are sent with its initial SIP INVITE message. Instead the media characteristics are sent by the called device in the Session Description Protocol (SDP) body of the next reliable message (200 OK) The called device s Offer will contain its IP Address, UDP Port number, list of codecs etc. The calling device selects which of the offered codecs it wishes to use for the call and returns its Answer in the SDP body of a reliable SIP response (ACK) The Answer also contains the IP address and UDP port number etc of the calling device Delayed Offer is a mandatory part of the SIP standard Many SPs prefer Early Offer Ordinarily, the Offer or Answer cannot be sent with 100 Trying or 180 Ringing as 1XX messages are unreliable (unacknowledged) This can be resolved using PRACK.. discussed later.. INVITE 100 Trying 180 Ringing 200 OK w/ SDP (Offer) ACK w/ SDP (Answer) INVITE 100 Trying 180 Ringing 200 OK w/ SDP (Offer) ACK w/ SDP (Answer) Two Way Media 90

SIP Trunk Signaling and Basic Operation SIP Messaging - Delayed and Early Offer Inbound SIP Delayed Offer to Outbound SIP Early offer So what happens when Unified CM receives an inbound call on a Delayed Offer Trunk and needs to onward route the call over a Early Offer Trunk? It does not have the calling device s media characteristics and it needs to send an Offer in SDP with the outbound INVITE Solution Insert a Media Termination Point (MTP) and use its media characteristics to create the Offer in SDP with the outbound INVITE SIP Delayed Offer SIP Early Offer INVITE 100 Trying 180 Ringing 200 OK w/ SDP (Offer) ACK w/ SDP (Answer) INVITE w/ SDP (MTP) 100 Trying 180 Ringing 200 OK w/ SDP (Answer) ACK Two Way Media MTP 91

SIP Trunk Signaling and Basic Operation SIP Messaging Enabling SIP Early Offer Method 1 SIP Trunk MTP Required Checkbox SIP Line SCCP Line SIP Trunk H323 Trunk MGCP Trunk MTP MTP MTP MTP MTP SIP Trunk with Early Offer SIP Trunk with Early Offer SIP Trunk with Early Offer SIP Trunk with Early Offer SIP Trunk with Early Offer Using the MTP Required option : SIP Early Offer Trunks use the Trunk s Media Termination Point (MTP) resources, inserting an MTP into the media path for every outbound call sending the MTP s IP Address, UDP port number and codec in the SDP body of the initial SIP INVITE instead of those of the endpoint. This has a number of disadvantages : MTPs support a single Audio codec only e.g. G711 or G729. The passthru codec is not supported excluding the use of SRTP and video calls. Since the Trunk s MTPs are used rather than the calling device s MTPs - The media path is forced to follow the signaling path. 92

SIP Trunk Signaling and Basic Operation SIP Messaging Enabling SIP Early Offer Method 2 New Features Cisco SIP Phones SIP Line SCCP Line SIP Trunk with Early Offer SIP Trunk with Early Offer New SIP Profile checkbox Early Offer support for voice and video calls (insert MTP if needed) Newer SCCP Phones MTP Older SCCP Phones SIP Early Offer SCCP Line SIP Trunk SIP Trunk SIP Trunk with Early Offer SIP Trunk with Early Offer SIP Trunk with Early Offer For Calls from trunks and devices that can provide their IP Address, UDP port number and supported codecs - This information is sent in the SDP body of the initial SIP Invite on the outbound Early Offer Trunk. No MTP is used for the Early Offer MTP SIP Delayed Offer MTP H323 Slow Start H323 Fast Start MGCP Gateway H323 Trunk H323 Trunk MGCP Trunk SIP Trunk with Early Offer SIP Trunk with Early Offer SIP Trunk with Early Offer For Calls from trunks and devices that cannot provide Early Offer information use the calling device s MTP resources (first) or the outbound trunk s MTPs (second) to create a SIP Offer for an unencrypted voice call. (SRTP and video can subsequently be initiated by the called device) 93

SIP Trunk Signaling and Basic Operation SIP Messaging Enabling SIP Early Offer Method 2 New Features Cisco SIP Phones SIP Line SIP Trunk with Early Offer MTP SIP Delayed Offer SIP Trunk SIP Trunk with Early Offer SCCP Line Newer SCCP Phones SIP Trunk with Early Offer MTP H323 Slow Start H323 Trunk SIP Trunk with Early Offer MTP Older SCCP Phones SCCP Line SIP Trunk with Early Offer H323 Fast Start H323 Trunk SIP Trunk with Early Offer SIP Trunk SIP Trunk with Early Offer MGCP Trunk SIP Trunk with Early Offer SIP Early Offer MGCP Gateway Benefits of Early Offer support for voice and video calls (insert MTP if needed) Reduced MTP usage Single voice codec MTP limitation removed (by using the pass through codec) Voice codecs sent in SIP Offer based on calling device capabilities & region settings Video Calls supported Encryption supported Use of the Calling device s MTP rather than Trunk s MTP - media does not have to follow the signaling path 94

SIP Trunk Signaling and Operation PRACK (1) SIP Early Media Using Provisional Acknowledgement (PRACK) SIP defines two types of responses: Final and Provisional. Final responses convey the result of the processed request, and are sent reliably (i.e. they are acknowledged). Provisional responses provide information on the progress of the request, but are not sent reliably so the sender of a provisional response does know that it has been received. To send an Offer or Answer with a provisional 1XX response these responses must be sent reliably.. PRACK Provisional Acknowledgement is used to provide 1XX responses with reliability. Diagram : Early Offer with Early Media INVITE w/ SDP Supported:100rel 100 Trying 183 Progress w/ SDP Require:100rel INVITE w/ SDP Supported:100rel 100 Trying 183 Progress w/ SDP Require:100rel PRACK 200 OK (PRACK) Two Way Media PRACK 200 OK (PRACK) 95

SIP Trunk Signaling and Operation PRACK (2) SIP Early Media Using Provisional Acknowledgement (PRACK) Like final responses, by using PRACK - 1XX messages will be periodically re-sent until their receipt is acknowledged by the receiver by sending a PRACK, which is also acknowledged by the 1XX sender. Using PRACK can reduce the number of SIP messages that need to be sent before two way media can be established PRACK is useful in situations where long Round Trip Times between SIP devices can cause a delay to media cut through or media clipping PRACK can be enabled on the SIP Trunk Profile by setting SIPRel1XX Options to enabled Diagram : Delayed Offer with Early Media INVITE Supported:100rel 100 Trying 183 Progress w/ SDP Require:100rel PRACK w/ SDP INVITE Supported:100rel 100 Trying 183 Progress w/ SDP Require:100rel PRACK w/ SDP 200 OK (PRACK) Two Way Media 200 OK (PRACK) 96

SIP Trunk Design Considerations Using standard Call Manager Groups and multiple destinations IP addresses A SIP Trunk D A SIP Trunk D B E B E C F C F Cluster 1 Cluster 2 Cluster 1 Cluster 2 Unified CM SIP Trunks will only accept inbound calls from a device with an IP address that has been defined as a destination IP address on the Trunk Cluster 1 SIP Trunk configuration Servers A, B and C in SIP Trunk s Call Manager Group Servers D, E and F are configured as Trunk destinations Cluster 2 SIP Trunk configuration Servers D, E and F in SIP Trunk s Call Manager Group Servers A, B and C are configured as Trunk destinations 97

SIP Trunk Design Considerations Using Run on all Active Unified CM Nodes and multiple destination IP addresses E E A SIP Trunk F A SIP Trunk F B G B G C H C H D I D I Cluster 1 Cluster 2 Cluster 1 Cluster 2 Unified CM SIP Trunks will only accept inbound calls from a device with an IP address that has been defined as a destination IP address on the Trunk Cluster 1 SIP Trunk configuration Cluster 2 SIP Trunk configuration The SIP Trunk has an active SIP daemon on Servers A, B, C and D Servers E, F, G, H and I are configured as Trunk destinations The SIP Trunk has an active SIP daemon on Servers E, F, G, H and I Servers A, B, C and D are configured as Trunk destinations 98

SIP Trunk Design Considerations Using standard Call Manager Groups, Run on all Active Unified CM Nodes and multiple destination IP addresses A A B SIP Trunk F B SIP Trunk F C G C G D H D H E E Cluster 1 Cluster 2 Cluster 1 Cluster 2 Unified CM SIP Trunks will only accept inbound calls from a device with an IP address that has been defined as a destination IP address on the Trunk Cluster 1 SIP Trunk configuration Cluster 2 SIP Trunk configuration The SIP Trunk has an active SIP daemon on Servers A, B, C, D and E Servers F, G and H are defined as Trunk destinations Servers F, G and H in SIP Trunk s Call Manager Group Servers A, B, C, D and E are defined as Trunk destinations 99

Signaling Delay SIP Delayed Offer Trunks Messages exchanged before the caller hears ringback tone One way signaling Delay INVITE 100 Trying 180 Ringing 200 OK w/ SDP (Offer) ACK w/ SDP (Answer) Delay before the caller hears the called user after ringback stops SME Two Way Media INVITE 100 Trying 180 Ringing 200 OK w/ SDP (Offer) ACK w/ SDP (Answer) Messages exchanged before called user hears the caller after picking up their handset 100

Signaling Delay SIP Early Offer Trunks Messages exchanged before the caller hears ringback tone One way signaling Delay INVITE w/ SDP (Offer) 100 Trying 180 Ringing 200 OK w/ SDP (Answer) ACK Two Way Media INVITE w/ SDP (Offer) 100 Trying 180 Ringing 200 OK w/ SDP (Answer) ACK Delay before the caller hears the called user after ringback stops Messages exchanged before called user hears the caller after picking up their handset 101